similar to: struggling with the "g" flag

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

I've got several agents on a queue. However, they often forget to go "not ready" or log off when they can't answer the phone. I would like a person calling my queue to be on the queue for a max of 2 minutes, and I'm using the rrmemory strategy. I put a timeout of 12 on the call to my agent in the [AgentQ] context (they log on using Agentlogincallback). It all seems to

Hi, Im getting daily segfaults when running 40-100 cuncurrent calls in G729 passthrough mode. Any thoughs on why this is happening is most appreciated. #0 0x0000003cd773356f in __strlen_sse42 () from /lib64/libc.so.6 #1 0x000000000043b352 in update_bridgepeer (c0=0x7faeac06a320, c1=0x7faea4711570) at channel.c:4791 #2 0x0000000000445f42 in ast_channel_bridge (c0=0x7faeac06a320,

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Well, sort of :) We have agents using the AgentLoginCallBack functionality. The agents log in using their agent number, with the extension automatically entered for them. When they log out, they again use the AgentLoginCallBack app, but using just a "#" for the new extension (logs them out). Occasionally, an Agent simply refuses to log out. You get a message "That Agent Is

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

We just had * crash on us - no calls could be made / received. We had to kill -9 the * process. Checking the error logs, I came across these two lines, with the times matching the crash: Aug 26 13:48:00 WARNING[19282] pbx.c: Local/6024@AgentQ-94ce,2 already has PBX structure?? Aug 26 13:48:00 WARNING[19282] channel.c: Thread -1105359952 Blocking 'Local/6024@AgentQ-94ce,2', already

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

Ist there any possibility to use the funktion "consultative transfer"? ( have 2 ISDN-pones attached to the hfc-nt card, configured as zap) With the "#"-key it ist possible to park the call or to make a "blind transfer" at the moment. I have activated threewaycalling in the zapata.conf file: ; internal S0 bus (first hfc/s card): context=local signalling =

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

Hi all, I'm trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn't see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any